Hello,
I have found asterisk is dialing a random number when it recieves a call,
would anyone know why? The first thing I noticed found peer 4563 (this is
a n Xlite Client)
Many thanks,
Spencer
SIP Debugging Enabled
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 10
Record-Route: <sip:82.70.154.145;ftag=as3606b893;lr=on>
Via: SIP/2.0/UDP 82.70.154.145;branch=z9hG4bK2922.6c170001.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: MSS VoIP Gateway
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 31661 31661 IN IP4 213.166.5.129
s=session
c=IN IP4 213.166.5.129
t=0 0
m=audio 14474 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 12 lines)---
Using latest request as basis request
Sending to 82.70.154.145 : 5060 (NAT)
Found peer '4563'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.129:14474
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 448715046363 in local-sip
list_route: hop: <sip:82.70.154.145;ftag=as3606b893;lr=on>
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Length: 0
---
-- Executing Dial("SIP/4563-5e36",
"SIP/[EMAIL PROTECTED]:5061|60|r")
We're at 192.168.4.3 port 35002
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.4.5:5061:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
To: <sip:[EMAIL PROTECTED]:5061>
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 8318 8318 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 35002 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called [EMAIL PROTECTED]:5061
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 100 Trying
To: <sip:[EMAIL PROTECTED]:5061>
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 180 Ringing
To: <sip:[EMAIL PROTECTED]:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 200 OK
To: <sip:[EMAIL PROTECTED]:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Contact: PSTN Line <sip:[EMAIL PROTECTED]:5061>
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
ontent-Type: application/sdp
v=0
o=- 3069797 3069797 IN IP4 192.168.4.5
s=-
c=IN IP4 192.168.4.5
t=0 0
m=audio 16452 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.4.5:16452
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[EMAIL PROTECTED]:5061>
set_destination: Parsing <sip:[EMAIL PROTECTED]:5061> for
address/port to send to
set_destination: set destination to 192.168.4.5, port 5061
Transmitting (no NAT) to 192.168.4.5:5061:
ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK660ef268
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
To: <sip:[EMAIL PROTECTED]:5061>;tag=d416591c6d2e2378i1
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>;tag=as7681341a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Length: 0
---
-- SIP/192.168.4.5:5061-05b4 is ringing
-- SIP/192.168.4.5:5061-05b4 answered SIP/4563-5e36
We're at 82.70.154.145 port 35040
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with capability 0x100 (g729)
Answering with capability 0x400 (ilbc)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
Record-Route: <sip:82.70.154.145;ftag=as3606b893;lr=on>
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>;tag=as7681341a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Type: application/sdp
Content-Length: 315
v=0
o=root 8318 8318 IN IP4 82.70.154.145
s=session
c=IN IP4 82.70.154.145
t=0 0
m=audio 35040 RTP/AVP 3 0 8 18 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Attempting native bridge of SIP/4563-5e36 and
SIP/192.168.4.5:5061-05b4
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 82.70.154.145;branch=0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK538adb90
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>;tag=as7681341a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: MSS VoIP Gateway
Content-Length: 0
P-hint: rr-enforced
--- (12 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
BYE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 82.70.154.145;branch=z9hG4bK3922.c51fd76.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK4eaf2728
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>;tag=as7681341a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: MSS VoIP Gateway
Content-Length: 0
Route: <sip:[EMAIL PROTECTED]:5061>
P-hint: rr-enforced
--- (13 headers 0 lines)---
Sending to 82.70.154.145 : 5060 (NAT)
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK3922.c51fd76.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK4eaf2728
From: "unknown" <sip:[EMAIL PROTECTED]>;tag=as3606b893
To: <sip:[EMAIL PROTECTED]>;tag=as7681341a
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Length: 0
---
set_destination: Parsing <sip:[EMAIL PROTECTED]:5061> for
address/port to send to
set_destination: set destination to 192.168.4.5, port 5061
Reliably Transmitting (no NAT) to 192.168.4.5:5061:
BYE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK2f906ce7
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
To: <sip:[EMAIL PROTECTED]:5061>;tag=d416591c6d2e2378i1
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
---
== Spawn extension (local-sip, 448715046363, 1) exited non-zero on
'SIP/4563-5e36'
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 200 OK
To: <sip:[EMAIL PROTECTED]:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:[EMAIL PROTECTED]:5061>;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK2f906ce7
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
spitfire*CLI> sip no debug
SIP Debugging Disabled
spitfire*CLI>
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
